From cf84d88dd68bc4c8fc52803832083c43a796b997 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Damian=20Kara=C5=9B?= <s176030@student.pg.edu.pl>
Date: Thu, 4 Nov 2021 19:03:49 +0100
Subject: [PATCH] :ALSA_support: buffers convertion for input
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Signed-off-by: Damian KaraĹ <s176030@student.pg.edu.pl>
---
src/cpp/ALSA_support.cpp | 183 ++++++++++++++++++++++++++-----------
src/include/ALSA_support.h | 6 +-
2 files changed, 132 insertions(+), 57 deletions(-)
diff --git a/src/cpp/ALSA_support.cpp b/src/cpp/ALSA_support.cpp
index 786a48c..236b1cc 100644
--- a/src/cpp/ALSA_support.cpp
+++ b/src/cpp/ALSA_support.cpp
@@ -350,75 +350,149 @@ int DSP::ALSA_object_t::open_alsa_device(snd_pcm_stream_t stream_type)
size_b = audio_inbuffer_size_in_frames * no_of_channels_alsa * no_of_bytes_in_channel;
- pcm_buffer.resize(DSP::NoOfAudioOutputBuffers);
- pcm_buffer_size_in_frames.resize(DSP::NoOfAudioOutputBuffers);
-
- switch (no_of_bytes_in_channel)
+ if (stream_type == SND_PCM_STREAM_PLAYBACK)
{
- case 1:
- buffers_8bit.resize(DSP::NoOfAudioOutputBuffers);
-
- for(unsigned int ind = 0; ind < DSP::NoOfAudioOutputBuffers; ind++)
- {
- buffers_8bit[ind].resize(size_b / no_of_bytes_in_channel);
- pcm_buffer[ind] = (uint8_t *)(buffers_8bit[ind].data());
- pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_8bit[ind].size() / no_of_channels_alsa;
- }
- break;
-
- case 2:
- buffers_16bit.resize(DSP::NoOfAudioOutputBuffers);
+ pcm_buffer.resize(DSP::NoOfAudioOutputBuffers);
+ pcm_buffer_size_in_frames.resize(DSP::NoOfAudioOutputBuffers);
+ switch (no_of_bytes_in_channel)
+ {
+ case 1:
+ buffers_8bit.resize(DSP::NoOfAudioOutputBuffers);
- for(unsigned int ind = 0; ind < DSP::NoOfAudioOutputBuffers; ind++)
- {
- buffers_16bit[ind].resize(size_b / no_of_bytes_in_channel);
- pcm_buffer[ind] = (uint8_t *)(buffers_16bit[ind].data());
- pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_16bit[ind].size() / no_of_channels_alsa; // buffer_size_in_frames should also be a vector since this value is not always used in this call and might be needed in the future
- }
- break;
+ for(unsigned int ind = 0; ind < DSP::NoOfAudioOutputBuffers; ind++)
+ {
+ buffers_8bit[ind].resize(size_b / no_of_bytes_in_channel);
+ pcm_buffer[ind] = (uint8_t *)(buffers_8bit[ind].data());
+ pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_8bit[ind].size() / no_of_channels_alsa;
+ }
+ break;
- case 3:
- case 4:
- if (IsHigherQualityMode)
- {
- buffers_32bit.resize(DSP::NoOfAudioOutputBuffers);
+ case 2:
+ buffers_16bit.resize(DSP::NoOfAudioOutputBuffers);
for(unsigned int ind = 0; ind < DSP::NoOfAudioOutputBuffers; ind++)
{
- buffers_32bit[ind].resize(size_b / no_of_bytes_in_channel);
- pcm_buffer[ind] = (uint8_t *)(buffers_32bit[ind].data());
- pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_32bit[ind].size() / no_of_channels_alsa;
+ buffers_16bit[ind].resize(size_b / no_of_bytes_in_channel);
+ pcm_buffer[ind] = (uint8_t *)(buffers_16bit[ind].data());
+ pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_16bit[ind].size() / no_of_channels_alsa; // buffer_size_in_frames should also be a vector since this value is not always used in this call and might be needed in the future
+ }
+ break;
+
+ case 3:
+ case 4:
+ if (IsHigherQualityMode)
+ {
+ buffers_32bit.resize(DSP::NoOfAudioOutputBuffers);
+
+ for(unsigned int ind = 0; ind < DSP::NoOfAudioOutputBuffers; ind++)
+ {
+ buffers_32bit[ind].resize(size_b / no_of_bytes_in_channel);
+ pcm_buffer[ind] = (uint8_t *)(buffers_32bit[ind].data());
+ pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_32bit[ind].size() / no_of_channels_alsa;
+ }
}
- }
- else //! native mode
- {
- buffers_32bit_f.resize(DSP::NoOfAudioOutputBuffers);
+ else //! native mode
+ {
+ buffers_32bit_f.resize(DSP::NoOfAudioOutputBuffers);
+
+ for(unsigned int ind = 0; ind < DSP::NoOfAudioOutputBuffers; ind++)
+ {
+ buffers_32bit_f[ind].resize(size_b / no_of_bytes_in_channel);
+ pcm_buffer[ind] = (uint8_t *)(buffers_32bit_f[ind].data());
+ pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_32bit_f[ind].size() / no_of_channels_alsa;
+ }
+ }
+ break;
+
+ case 8:
+ buffers_64bit.resize(DSP::NoOfAudioOutputBuffers);
for(unsigned int ind = 0; ind < DSP::NoOfAudioOutputBuffers; ind++)
{
- buffers_32bit_f[ind].resize(size_b / no_of_bytes_in_channel);
- pcm_buffer[ind] = (uint8_t *)(buffers_32bit_f[ind].data());
- pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_32bit_f[ind].size() / no_of_channels_alsa;
+ buffers_64bit[ind].resize(size_b / no_of_bytes_in_channel);
+ pcm_buffer[ind] = (uint8_t *)(buffers_64bit[ind].data());
+ pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_64bit[ind].size() / no_of_channels_alsa;
}
- }
- break;
+ break;
- case 8:
- buffers_64bit.resize(DSP::NoOfAudioOutputBuffers);
+ default:
+ DSP::log << "Unsupported no of bytes in channel" << endl;
+ return -1;
+ break;
+ }
+ }
+ else // (stream_type == SND_PCM_STREAM_CAPTURE)
+ {
+ pcm_buffer.resize(DSP::NoOfAudioInputBuffers);
+ pcm_buffer_size_in_frames.resize(DSP::NoOfAudioInputBuffers);
+ switch (no_of_bytes_in_channel)
+ {
+ case 1:
+ buffers_8bit.resize(DSP::NoOfAudioInputBuffers);
- for(unsigned int ind = 0; ind < DSP::NoOfAudioOutputBuffers; ind++)
- {
- buffers_64bit[ind].resize(size_b / no_of_bytes_in_channel);
- pcm_buffer[ind] = (uint8_t *)(buffers_64bit[ind].data());
- pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_64bit[ind].size() / no_of_channels_alsa;
- }
- break;
+ for(unsigned int ind = 0; ind < DSP::NoOfAudioInputBuffers; ind++)
+ {
+ buffers_8bit[ind].resize(size_b / no_of_bytes_in_channel);
+ pcm_buffer[ind] = (uint8_t *)(buffers_8bit[ind].data());
+ pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_8bit[ind].size() / no_of_channels_alsa;
+ }
+ break;
- default:
- DSP::log << "Unsupported no of bytes in channel" << endl;
- return -1;
- break;
+ case 2:
+ buffers_16bit.resize(DSP::NoOfAudioInputBuffers);
+
+ for(unsigned int ind = 0; ind < DSP::NoOfAudioInputBuffers; ind++)
+ {
+ buffers_16bit[ind].resize(size_b / no_of_bytes_in_channel);
+ pcm_buffer[ind] = (uint8_t *)(buffers_16bit[ind].data());
+ pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_16bit[ind].size() / no_of_channels_alsa; // buffer_size_in_frames should also be a vector since this value is not always used in this call and might be needed in the future
+ }
+ break;
+
+ case 3:
+ case 4:
+ if (IsHigherQualityMode)
+ {
+ buffers_32bit.resize(DSP::NoOfAudioInputBuffers);
+
+ for(unsigned int ind = 0; ind < DSP::NoOfAudioInputBuffers; ind++)
+ {
+ buffers_32bit[ind].resize(size_b / no_of_bytes_in_channel);
+ pcm_buffer[ind] = (uint8_t *)(buffers_32bit[ind].data());
+ pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_32bit[ind].size() / no_of_channels_alsa;
+ }
+ }
+
+ else //! native mode
+ {
+ buffers_32bit_f.resize(DSP::NoOfAudioInputBuffers);
+
+ for(unsigned int ind = 0; ind < DSP::NoOfAudioInputBuffers; ind++)
+ {
+ buffers_32bit_f[ind].resize(size_b / no_of_bytes_in_channel);
+ pcm_buffer[ind] = (uint8_t *)(buffers_32bit_f[ind].data());
+ pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_32bit_f[ind].size() / no_of_channels_alsa;
+ }
+ }
+ break;
+
+ case 8:
+ buffers_64bit.resize(DSP::NoOfAudioInputBuffers);
+
+ for(unsigned int ind = 0; ind < DSP::NoOfAudioInputBuffers; ind++)
+ {
+ buffers_64bit[ind].resize(size_b / no_of_bytes_in_channel);
+ pcm_buffer[ind] = (uint8_t *)(buffers_64bit[ind].data());
+ pcm_buffer_size_in_frames[ind] = (snd_pcm_sframes_t) buffers_64bit[ind].size() / no_of_channels_alsa;
+ }
+ break;
+
+ default:
+ DSP::log << "Unsupported no of bytes in channel" << endl;
+ return -1;
+ break;
+ }
}
if (blocking_mode == false)
@@ -435,6 +509,7 @@ int DSP::ALSA_object_t::open_alsa_device(snd_pcm_stream_t stream_type)
else
{
rc = snd_pcm_nonblock(alsa_handle, 1);
+
if (rc < 0)
{
DSP::log << "Unable to set non blocking mode" << endl;
diff --git a/src/include/ALSA_support.h b/src/include/ALSA_support.h
index 60199be..90d1e97 100644
--- a/src/include/ALSA_support.h
+++ b/src/include/ALSA_support.h
@@ -79,9 +79,9 @@ namespace DSP {
/***************************************************/
/****************AUDIO INPUT TEST*******************/
- DSP::e::SampleType InSampleType;
- static void CALLBACK waveInProc(HWAVEIN hwi, UINT uMsg, uint32_t dwInstance, uint32_t dwParam1, uint32_t dwParam2);
- std::vector<std::vector<int8_t>> input_buffers;
+ //DSP::e::SampleType InSampleType;
+ //static void CALLBACK waveInProc(HWAVEIN hwi, UINT uMsg, uint32_t dwInstance, uint32_t dwParam1, uint32_t dwParam2);
+ //std::vector<std::vector<int8_t>> input_buffers;
/**********************END**************************/
/***************************************************/
--
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